Contact emailsorp...@chromium.org Specification https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-codec
Summary This new API extends WebRTC encoding parameters to allow developers to choose a specific negotiated codec to be used for encoding an RTP stream. Blink componentBlink>WebRTC>PeerConnection <https://bugs.chromium.org/p/chromium/issues/list?q=component:Blink%3EWebRTC%3EPeerConnection> Motivation We want to be able to change codec with RTCRtpSender.setParameters() in order to do the following: - Allow different codecs on different encodings. - Make it possible to change codec without re-negotiating. - Allow specifying both codec and scalabilityMode with a single API call. Current users of the WebRTC APIs can achieve the similar functionality with multiple calls that are not quite easy to use (renegotiate with a different codec order in the SDP, which is a heavy operation) and are not as efficient (changing the codec and then changing the scalability mode or other parameters may add an extra I-frame that is not needed). This API also allows to have mixed-codec simulcast, which was not possible previously. TAG reviewhttps://github.com/w3ctag/design-reviews/issues/836 TAG review statusPending Risks Interoperability and Compatibility Interoperability risks are low. This is a new dictionary member and API surface, if it isn't used by current applications, it should not cause any impact on compatibility. *Gecko*: No signal ( https://github.com/mozilla/standards-positions/issues/789) *WebKit*: No signal ( https://github.com/WebKit/standards-positions/issues/179) *Web developers*: No signals *Other signals*: Ergonomics No ergonomic risks. This API is part of WebRTC and will be used with other encoding parameters. One of its goal is to improve the WebRTC ergonomics by removing the need to make several calls with possible side effects to change the active codec, which should improve performance as well. Debuggability WebRTC is not supported by DevTools at the moment. chrome://webrtc-internals will reflect in the RTP stream statistics which codec is currently used. Is this feature fully tested by web-platform-tests <https://chromium.googlesource.com/chromium/src/+/main/docs/testing/web_platform_tests.md> ?Yes Requires code in //chrome?False Tracking bughttps://bugs.chromium.org/p/chromium/issues/detail?id=1442194 Estimated milestones No milestones specified Link to entry on the Chrome Platform Status https://chromestatus.com/feature/5200982281027584 This intent message was generated by Chrome Platform Status <https://chromestatus.com/>. -- You received this message because you are subscribed to the Google Groups "blink-dev" group. To unsubscribe from this group and stop receiving emails from it, send an email to blink-dev+unsubscr...@chromium.org. To view this discussion on the web visit https://groups.google.com/a/chromium.org/d/msgid/blink-dev/CADRnnSXEi2i3DzkXQo1ptX-KR%2BULh-4LOt5zpOi14BTstZN_9g%40mail.gmail.com.
